FreePBX 183 Session Progress (SDP)

Eugen · September 5, 2024, 1:09pm

Good day.
I need help. I am setting up my FreePBX connection to a SIP provider. Everything works fine, but the provider requires that my server not send them 183 Session Progress (SDP).
183

I have already done Edit Trunk → Inband Progress → No and Inbound Routes → Signal RINGING → Yes. This enabled 180 Ringing for me (which the provider also required), but did not get rid of 183 Session Progress (SDP). I have already tried many settings related to early media and answers, but this did not give results. However, I get the correct result if I simply do

[from-pstn-custom]
exten => _X .,1,Dial(PJSIP/1000)

I just can’t imagine what in FreePBC makes it send the 183 code.

david55 · September 5, 2024, 1:56pm

Probably the result of receiving 183 from the B side. If that is the reason, the only way to stop it is probably to use a dial string that calls multiple destinations, and have Asterisk send Ringing without having received it.

Better solution is to find a SIP provider. Normally the issue with 183 plus early media is that the early media wouldn’t be forwarded, by the provider, but the call would work if the B side succeeded.

Is the B side something other than a local phone?

A more certain way would be to answer the call immediately, but that could start billing for the callee, even though the call, eventually,. fails.

BlazeStudios · September 5, 2024, 2:12pm

Where does the Inbound Route for this DID send the call in the PBX? Is it sending it directly to an extension or something else?

Eugen · September 5, 2024, 2:26pm

That’s how it was set up originally. But the provider didn’t want to receive 200 OK before the phone was actually picked up.
And yes, side B is the local phone.

Eugen · September 5, 2024, 2:28pm

The call is routed directly to an extension that is registered to the telephone on my desk.

BlazeStudios · September 5, 2024, 2:32pm

Why does the phone matter here? This is a PBX there are numerous ways for this call to be answered before it gets sent to phone. If the call goes to voicemail, it’s going to return a 200 OK without the phone answering. A PBX can do a lot with a call and never once send the call to the phones that are connected to it.

BlazeStudios · September 5, 2024, 2:38pm

Then we are going to need to see more than what you have shown. You’ll need to log into the shell and do asterisk -r to get into the Asterisk CLI then do pjsip set logger on and make an inbound call that produces a 183 reply to the carrier. We need to see what the phone is sending back to Asterisk and what Asterisk is sending back to the provider.

Also this:

Ignores anything the PBX does with call handling such as adding dial options. We need to see what happens when a call is made and it goes through all the proper call handling in the PBX.

Eugen · September 5, 2024, 2:45pm

The phone is not matter here. I’m just trying to add details. And I know there are many ways to get 200 OK, but I don’t know how to stop getting 183.

Ok, I will give this information. But unfortunately only tomorrow, so I am not in the office today and do not have access.

Eugen · September 6, 2024, 5:26am

<--- Received SIP request (417 bytes) from UDP:10.9.10.9:5060 --->
OPTIONS sip:10.9.198.144:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.10.9:5060;branch=z9hG4bKiei0cx0xiss11eiqvv4cwxv4e;Role=3;Hpt=8f58_16;pth=0;X-HwDim=4
Call-ID: [email protected]
From: <sip:[email protected]>;tag=12lh22ye
To: <sip:10.9.198.144>
CSeq: 1 OPTIONS
Contact: <sip:10.9.10.9:36696;transport=udp;Hpt=8f58_16>;expires=65535
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0


<--- Transmitting SIP response (865 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKiei0cx0xiss11eiqvv4cwxv4e;Role=3;Hpt=8f58_16;pth=0;X-HwDim=4
Call-ID: [email protected]
From: <sip:[email protected]>;tag=12lh22ye
To: <sip:10.9.198.144>;tag=z9hG4bKiei0cx0xiss11eiqvv4cwxv4e
CSeq: 1 OPTIONS
Accept: application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: FPBX-15.0.37.5(16.30.0)
Content-Length:  0

<--- Received SIP request (1199 bytes) from UDP:10.9.10.9:5060 --->
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.9.10.9:5060;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955"<sip:[email protected];transport=udp;user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952"<sip:[email protected];transport=udp;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Contact: <sip:[email protected]:5060;transport=udp;user=phone;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff>
Max-Forwards: 69
Supported: 100rel,timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: <tel:+4822669955;cpc=ordinary>
P-Early-Media: supported
Content-Length: 249
Content-Type: application/sdp

v=0
o=- 1628885996 1628885997 IN IP4 10.9.10.10
s=SBC call
c=IN IP4 10.9.10.10
t=0 0
m=audio 58374 RTP/AVP 8 0 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
a=3gOoBTC

<--- Transmitting SIP response (556 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>
CSeq: 1 INVITE
Server: FPBX-15.0.37.5(16.30.0)
Content-Length:  0


<--- Transmitting SIP response (743 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 INVITE
Server: FPBX-15.0.37.5(16.30.0)
Contact: <sip:10.9.198.144:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Transmitting SIP response (1090 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 INVITE
Server: FPBX-15.0.37.5(16.30.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.9.198.144:5060>
Content-Type: application/sdp
Content-Length:   304

v=0
o=- 1628885996 1628885999 IN IP4 10.9.10.10
s=Asterisk
c=IN IP4 10.9.10.10
t=0 0
m=audio 10466 RTP/AVP 8 0 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1090 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 INVITE
Server: FPBX-15.0.37.5(16.30.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.9.198.144:5060>
Content-Type: application/sdp
Content-Length:   304

v=0
o=- 1628885996 1628885999 IN IP4 10.9.10.10
s=Asterisk
c=IN IP4 10.9.10.10
t=0 0
m=audio 10466 RTP/AVP 8 0 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (1155 bytes) to UDP:192.168.144.31:5866 --->
INVITE sip:[email protected]:5866 SIP/2.0
Via: SIP/2.0/UDP 192.168.144.2:55060;rport;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>
Contact: <sip:[email protected]:55060>
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "+4822669955" <sip:[email protected]>
Max-Forwards: 70
User-Agent: FPBX-15.0.37.5(16.30.0)
Content-Type: application/sdp
Content-Length:   386

v=0
o=- 478029021 478029021 IN IP4 192.168.144.2
s=Asterisk
c=IN IP4 192.168.144.2
t=0 0
m=audio 11604 RTP/AVP 8 0 18 3 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (426 bytes) from UDP:192.168.144.31:5866 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.144.2:55060;rport=55060;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (556 bytes) from UDP:192.168.144.31:5866 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.144.2:55060;rport=55060;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>;tag=561630892
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 INVITE
Contact: <sip:[email protected]:5866>
User-Agent: Fanvil X3S 2.14.0.7387 0c383e224f3b
Allow-Events: talk,hold
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (1090 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 INVITE
Server: FPBX-15.0.37.5(16.30.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.9.198.144:5060>
Content-Type: application/sdp
Content-Length:   304

v=0
o=- 1628885996 1628885999 IN IP4 10.9.10.10
s=Asterisk
c=IN IP4 10.9.10.10
t=0 0
m=audio 10466 RTP/AVP 8 0 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


<--- Received SIP request (514 bytes) from UDP:10.9.10.9:5060 --->
CANCEL sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.9.10.9:5060;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Call-ID: [email protected]
From: "+4822669955"<sip:[email protected];transport=udp;user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952"<sip:[email protected];transport=udp;user=phone>
CSeq: 1 CANCEL
Max-Forwards: 69
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0


<--- Transmitting SIP response (476 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 CANCEL
Server: FPBX-15.0.37.5(16.30.0)
Content-Length:  0


<--- Transmitting SIP response (720 bytes) to UDP:10.9.10.9:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.9.10.9:5060;rport=5060;received=10.9.10.9;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Record-Route: <sip:10.9.10.9:5060;transport=udp;lr;Hpt=8ee8_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=13815>
Call-ID: [email protected]
From: "+4822669955" <sip:[email protected];user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952" <sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 INVITE
Server: FPBX-15.0.37.5(16.30.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Transmitting SIP request (438 bytes) to UDP:192.168.144.31:5866 --->
CANCEL sip:[email protected]:5866 SIP/2.0
Via: SIP/2.0/UDP 192.168.144.2:55060;rport;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 CANCEL
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-15.0.37.5(16.30.0)
Content-Length:  0


<--- Received SIP request (483 bytes) from UDP:10.9.10.9:5060 --->
ACK sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.9.10.9:5060;branch=z9hG4bKnpjnvzfz5vzjlp0okpjpnfk5a;Role=3;Hpt=8ee8_16
Call-ID: [email protected]
From: "+4822669955"<sip:[email protected];transport=udp;user=phone;cpc-rus=1>;tag=iumimmie-CC-1005-OFC-665
To: "+4822669952"<sip:[email protected];user=phone>;tag=318246fd-86bf-4bb2-bcb6-ccecc2b835cd
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<--- Received SIP response (398 bytes) from UDP:192.168.144.31:5866 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.2:55060;rport=55060;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>;tag=561630892
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 CANCEL
User-Agent: Fanvil X3S 2.14.0.7387 0c383e224f3b
Content-Length: 0


<--- Received SIP response (501 bytes) from UDP:192.168.144.31:5866 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.144.2:55060;rport=55060;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>;tag=561630892
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 INVITE
User-Agent: Fanvil X3S 2.14.0.7387 0c383e224f3b
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (422 bytes) to UDP:192.168.144.31:5866 --->
ACK sip:[email protected]:5866 SIP/2.0
Via: SIP/2.0/UDP 192.168.144.2:55060;rport;branch=z9hG4bKPj7e0b652d-5cec-4c29-867f-e057e44d3f39
From: "+4822669955" <sip:[email protected]>;tag=393aa43a-5208-403c-97eb-bd049e40cb11
To: <sip:[email protected]>;tag=561630892
Call-ID: aa4313c6-6c22-421c-8a7a-4db1522c6ca5
CSeq: 5282 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.37.5(16.30.0)
Content-Length:  0

david55 · September 6, 2024, 11:13am

Please repeat with verbosity at least 3, so we can see what, if anything, in the dial plan, triggers this, and also please use the full log file, not the console, so we can see the times of the events.

BlazeStudios · September 6, 2024, 12:26pm

So why is the reason you have to not have a 183 reply? Why can’t you have early media playing back? Because according to the provider’s INVITE they are sending the PBX:

P-Early-Media: supported

They indicate that early media is supported.

Eugen · September 6, 2024, 1:23pm

I thought about it. But the provider disagrees with me. And I’m not sure enough that this is the reason to convince him otherwise.
Do you think that FPBX will stop sending 183 if the invite does not support early media?

BlazeStudios · September 6, 2024, 1:35pm

We still need to see more verbose logging of a call.

What is the actual issue you are having with the provider by sending 183 replies?

Eugen · September 6, 2024, 2:26pm

Will have to postpone until Monday.

And the only problem is that the provider requires that there be no 183 replies. Everything works fine as is.

BlazeStudios · September 6, 2024, 2:31pm

That is a very strange requirement for a provider. For me, that would raise a flag and have me looking elsewhere for services.

Eugen · September 6, 2024, 3:40pm

I think the problem is not with the provider, but with the person with whom I am setting up. In response to my comments, he sends me call dumps from other clients and says “do the same”. We have been working with this provider for many years on PRI. But now they are switching everyone to SIP. And we need to keep our 300 numbers.

Eugen · September 9, 2024, 10:32am

@BlazeStudios, @david55
verbose logging

    -- Executing [+4822669952@from-pstn:1] Set("PJSIP/Provider_SIP_Trunk-00000191", "__DIRECTION=INBOUND") in new stack
    -- Executing [+4822669952@from-pstn:2] Gosub("PJSIP/Provider_SIP_Trunk-00000191", "sub-record-check,s,1(in,+4822669952,never)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("PJSIP/Provider_SIP_Trunk-00000191", "NOW=1725876672") in new stack
    -- Executing [s@sub-record-check:4] Set("PJSIP/Provider_SIP_Trunk-00000191", "__DAY=09") in new stack
    -- Executing [s@sub-record-check:5] Set("PJSIP/Provider_SIP_Trunk-00000191", "__MONTH=09") in new stack
    -- Executing [s@sub-record-check:6] Set("PJSIP/Provider_SIP_Trunk-00000191", "__YEAR=2024") in new stack
    -- Executing [s@sub-record-check:7] Set("PJSIP/Provider_SIP_Trunk-00000191", "__TIMESTR=20240909-131112") in new stack
    -- Executing [s@sub-record-check:8] Set("PJSIP/Provider_SIP_Trunk-00000191", "__FROMEXTEN=unknown") in new stack
    -- Executing [s@sub-record-check:9] Set("PJSIP/Provider_SIP_Trunk-00000191", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("PJSIP/Provider_SIP_Trunk-00000191", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "2?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?sub-record-check,in,1") in new stack
    -- Goto (sub-record-check,in,1)
    -- Executing [in@sub-record-check:1] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Inbound Recording Check to +4822669952") in new stack
    -- Executing [in@sub-record-check:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "FROMEXTEN=unknown") in new stack
    -- Executing [in@sub-record-check:3] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "13?Set(FROMEXTEN=+4822669955)") in new stack
    -- Executing [in@sub-record-check:4] Gosub("PJSIP/Provider_SIP_Trunk-00000191", "recordcheck,1(never,in,+4822669952)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Starting recording check against never") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/Provider_SIP_Trunk-00000191", "never") in new stack
    -- Goto (sub-record-check,recordcheck,14)
    -- Executing [recordcheck@sub-record-check:14] Set("PJSIP/Provider_SIP_Trunk-00000191", "__REC_POLICY_MODE=NEVER") in new stack
    -- Executing [recordcheck@sub-record-check:15] Goto("PJSIP/Provider_SIP_Trunk-00000191", "stoprec") in new stack
    -- Goto (sub-record-check,recordcheck,26)
    -- Executing [recordcheck@sub-record-check:26] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Stopping recording: in, +4822669952") in new stack
    -- Executing [recordcheck@sub-record-check:27] Set("PJSIP/Provider_SIP_Trunk-00000191", "__REC_STATUS=STOPPED") in new stack
    -- Executing [recordcheck@sub-record-check:28] System("PJSIP/Provider_SIP_Trunk-00000191", "/var/lib/asterisk/bin/stoprecording.php "PJSIP/Provider_SIP_Trunk-00000191"") in new stack
    -- Executing [recordcheck@sub-record-check:29] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [in@sub-record-check:5] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [+4822669952@from-pstn:3] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANNEL(tonezone)=en") in new stack
    -- Executing [+4822669952@from-pstn:4] Set("PJSIP/Provider_SIP_Trunk-00000191", "__FROM_DID=+4822669952") in new stack
    -- Executing [+4822669952@from-pstn:5] Set("PJSIP/Provider_SIP_Trunk-00000191", "returnhere=1") in new stack
    -- Executing [+4822669952@from-pstn:6] Gosub("PJSIP/Provider_SIP_Trunk-00000191", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [+4822669952@from-pstn:7] Set("PJSIP/Provider_SIP_Trunk-00000191", "CDR(did)=+4822669952") in new stack
    -- Executing [+4822669952@from-pstn:8] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?") in new stack
    -- Executing [+4822669952@from-pstn:9] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0 ?Set(CALLERID(name)=+4822669955)") in new stack
    -- Executing [+4822669952@from-pstn:10] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANNEL(musicclass)=none") in new stack
    -- Executing [+4822669952@from-pstn:11] Set("PJSIP/Provider_SIP_Trunk-00000191", "__MOHCLASS=none") in new stack
    -- Executing [+4822669952@from-pstn:12] Ringing("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [+4822669952@from-pstn:13] Set("PJSIP/Provider_SIP_Trunk-00000191", "__RINGINGSENT=TRUE") in new stack
    -- Executing [+4822669952@from-pstn:14] Set("PJSIP/Provider_SIP_Trunk-00000191", "__REVERSAL_REJECT=FALSE") in new stack
    -- Executing [+4822669952@from-pstn:15] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?post-reverse-charge") in new stack
    -- Goto (from-pstn,+4822669952,17)
    -- Executing [+4822669952@from-pstn:17] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [+4822669952@from-pstn:18] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack
    -- Executing [+4822669952@from-pstn:19] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack
    -- Executing [+4822669952@from-pstn:20] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLERID(name-pres)=allowed_not_screened") in new stack
    -- Executing [+4822669952@from-pstn:21] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLERID(num-pres)=allowed_not_screened") in new stack
    -- Executing [+4822669952@from-pstn:22] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "CallerID Entry Point") in new stack
    -- Executing [+4822669952@from-pstn:23] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CRM_DIRECTION=INBOUND") in new stack
    -- Executing [+4822669952@from-pstn:24] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CRM_SOURCE=+4822669955") in new stack
    -- Executing [+4822669952@from-pstn:25] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CRM_LINKEDID=1725876672.805") in new stack
    -- Executing [+4822669952@from-pstn:26] AGI("PJSIP/Provider_SIP_Trunk-00000191", "agi://127.0.0.1/sangomacrm.agi,true") in new stack
    -- <PJSIP/Provider_SIP_Trunk-00000191>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
    -- Executing [+4822669952@from-pstn:27] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
    -- Executing [+4822669952@from-pstn:28] Goto("PJSIP/Provider_SIP_Trunk-00000191", "from-did-direct,2440,1") in new stack
    -- Goto (from-did-direct,2440,1)
    -- Executing [2440@from-did-direct:1] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?ext-local,*2440,1") in new stack
    -- Executing [2440@from-did-direct:2] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?ext-local,2440,1:followme-check,2440,1") in new stack
    -- Goto (ext-local,2440,1)
    -- Executing [2440@ext-local:1] Set("PJSIP/Provider_SIP_Trunk-00000191", "__RINGTIMER=90") in new stack
    -- Executing [2440@ext-local:2] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(__CWIGNORE=)") in new stack
    -- Executing [2440@ext-local:3] Macro("PJSIP/Provider_SIP_Trunk-00000191", "exten-vm,novm,2440,1,1,1") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("PJSIP/Provider_SIP_Trunk-00000191", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/Provider_SIP_Trunk-00000191", "TOUCH_MONITOR=1725876672.805") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANCONTEXT=") in new stack
    -- Executing [s@macro-user-callerid:3] Progress("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [s@macro-user-callerid:4] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANCONTEXT=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANEXTENCONTEXT=Provider_SIP_Trunk-00000191") in new stack
    -- Executing [s@macro-user-callerid:6] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANEXTEN=Provider_SIP_Trunk-00000191") in new stack
       > 0x7fa1a8141170 -- Strict RTP learning after remote address set to: 10.9.10.10:25600
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLERID(number)=+4822669955") in new stack
    -- Executing [s@macro-user-callerid:8] Set("PJSIP/Provider_SIP_Trunk-00000191", "AMPUSER=+4822669955") in new stack
    -- Executing [s@macro-user-callerid:9] Set("PJSIP/Provider_SIP_Trunk-00000191", "HOTDESCKCHAN=Provider_SIP_Trunk-00000191") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/Provider_SIP_Trunk-00000191", "HOTDESKEXTEN=Provider_SIP_Trunk") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/Provider_SIP_Trunk-00000191", "HOTDESKCALL=0") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(HOTDESKCALL=1)") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?report") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "1?Set(REALCALLERIDNUM=+4822669955)") in new stack
    -- Executing [s@macro-user-callerid:16] Set("PJSIP/Provider_SIP_Trunk-00000191", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:18] Set("PJSIP/Provider_SIP_Trunk-00000191", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:19] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:20] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?report") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s@macro-user-callerid:29] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Macro Depth is 2") in new stack
    -- Executing [s@macro-user-callerid:30] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,31)
    -- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:32] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:33] Set("PJSIP/Provider_SIP_Trunk-00000191", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:34] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,50)
    -- Executing [s@macro-user-callerid:50] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLERID(number)=+4822669955") in new stack
    -- Executing [s@macro-user-callerid:51] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLERID(name)=+4822669955") in new stack
    -- Executing [s@macro-user-callerid:52] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:53] Set("PJSIP/Provider_SIP_Trunk-00000191", "CDR(cnam)=+4822669955") in new stack
    -- Executing [s@macro-user-callerid:54] Set("PJSIP/Provider_SIP_Trunk-00000191", "CDR(cnum)=+4822669955") in new stack
    -- Executing [s@macro-user-callerid:55] Set("PJSIP/Provider_SIP_Trunk-00000191", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("PJSIP/Provider_SIP_Trunk-00000191", "__EXTTOCALL=2440") in new stack
    -- Executing [s@macro-exten-vm:4] Set("PJSIP/Provider_SIP_Trunk-00000191", "__PICKUPMARK=2440") in new stack
    -- Executing [s@macro-exten-vm:5] Set("PJSIP/Provider_SIP_Trunk-00000191", "RT=90") in new stack
    -- Executing [s@macro-exten-vm:6] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
    -- Executing [s@macro-exten-vm:7] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:8] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Gosub(ext-intercom,*802440,1())") in new stack
    -- Executing [s@macro-exten-vm:9] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:10] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?ChanSpy(PJSIP/2440,q)") in new stack
    -- Executing [s@macro-exten-vm:11] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:12] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
    -- Executing [s@macro-exten-vm:13] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:14] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Gosub(ext-intercom,*802440,1())") in new stack
    -- Executing [s@macro-exten-vm:15] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:16] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?ChanSpy(PJSIP/2440,q)") in new stack
    -- Executing [s@macro-exten-vm:17] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:18] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?startcheck:exitcheck") in new stack
    -- Goto (macro-exten-vm,s,19)
    -- Executing [s@macro-exten-vm:19] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?featureSIP:featurePJSIP") in new stack
    -- Goto (macro-exten-vm,s,23)
    -- Executing [s@macro-exten-vm:23] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
    -- Executing [s@macro-exten-vm:24] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:25] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?featuremoniPJSIP:featuremoniSIP") in new stack
    -- Goto (macro-exten-vm,s,26)
    -- Executing [s@macro-exten-vm:26] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?ChanSpy(PJSIP/2440,q)") in new stack
    -- Executing [s@macro-exten-vm:27] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:28] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?check-ext-intercom:featuremoniSIP") in new stack
    -- Goto (macro-exten-vm,s,31)
    -- Executing [s@macro-exten-vm:31] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?ext-intercomSIP:ext-intercomPJSIP") in new stack
    -- Goto (macro-exten-vm,s,32)
    -- Executing [s@macro-exten-vm:32] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Gosub(ext-intercom,*802440,1())") in new stack
    -- Executing [s@macro-exten-vm:33] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:34] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?exitcheck:ext-intercomSIP") in new stack
    -- Goto (macro-exten-vm,s,37)
    -- Executing [s@macro-exten-vm:37] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Exiting Checks") in new stack
    -- Executing [s@macro-exten-vm:38] Gosub("PJSIP/Provider_SIP_Trunk-00000191", "sub-record-check,s,1(exten,2440,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "13?initialized") in new stack
    -- Goto (sub-record-check,s,10)
    -- Executing [s@sub-record-check:10] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("PJSIP/Provider_SIP_Trunk-00000191", "REC_POLICY_MODE_SAVE=NEVER") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Exten Recording Check between +4822669955 and 2440") in new stack
    -- Executing [exten@sub-record-check:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLTYPE=external") in new stack
    -- Executing [exten@sub-record-check:3] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(CALLTYPE=)") in new stack
    -- Executing [exten@sub-record-check:4] Set("PJSIP/Provider_SIP_Trunk-00000191", "CALLEE=yes") in new stack
    -- Executing [exten@sub-record-check:5] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:6] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?callee") in new stack
    -- Goto (sub-record-check,exten,11)
    -- Executing [exten@sub-record-check:11] Gosub("PJSIP/Provider_SIP_Trunk-00000191", "recordcheck,1(yes,external,2440)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Starting recording check against yes") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/Provider_SIP_Trunk-00000191", "yes") in new stack
    -- Goto (sub-record-check,recordcheck,9)
    -- Executing [recordcheck@sub-record-check:9] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "1?Return()") in new stack
    -- Executing [exten@sub-record-check:12] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [s@macro-exten-vm:39] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,45)
    -- Executing [s@macro-exten-vm:45] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "0?clrheader,1()") in new stack
    -- Executing [s@macro-exten-vm:46] Macro("PJSIP/Provider_SIP_Trunk-00000191", "dial-one,90,HhTtr,2440") in new stack
    -- Executing [s@macro-dial-one:1] Set("PJSIP/Provider_SIP_Trunk-00000191", "DEXTEN=2440") in new stack
    -- Executing [s@macro-dial-one:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CRM_SOURCE=+4822669955") in new stack
    -- Executing [s@macro-dial-one:3] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(__EXTTOCALL=2440)") in new stack
    -- Executing [s@macro-dial-one:4] Set("PJSIP/Provider_SIP_Trunk-00000191", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:5] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:6] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:7] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,10)
    -- Executing [s@macro-dial-one:10] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?continue") in new stack
    -- Executing [s@macro-dial-one:12] Set("PJSIP/Provider_SIP_Trunk-00000191", "EXTHASCW=") in new stack
    -- Executing [s@macro-dial-one:13] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,14)
    -- Executing [s@macro-dial-one:14] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?docfu:skip3") in new stack
    -- Goto (macro-dial-one,s,18)
    -- Executing [s@macro-dial-one:18] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?next2:continue") in new stack
    -- Goto (macro-dial-one,s,19)
    -- Executing [s@macro-dial-one:19] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?continue") in new stack
    -- Goto (macro-dial-one,s,27)
    -- Executing [s@macro-dial-one:27] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("PJSIP/Provider_SIP_Trunk-00000191", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("PJSIP/Provider_SIP_Trunk-00000191", "DEVICES=2440") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(DEVICES=440)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("PJSIP/Provider_SIP_Trunk-00000191", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("PJSIP/Provider_SIP_Trunk-00000191", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("PJSIP/Provider_SIP_Trunk-00000191", "THISDIAL=PJSIP/2440") in new stack
    -- Executing [dstring@macro-dial-one:8] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?docheck") in new stack
    -- Executing [dstring@macro-dial-one:9] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Debug: Found PJSIP Destination PJSIP/2440") in new stack
    -- Executing [dstring@macro-dial-one:10] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?doset") in new stack
    -- Executing [dstring@macro-dial-one:11] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("PJSIP/Provider_SIP_Trunk-00000191", "THISDIAL=PJSIP/2440/sip:[email protected]:5953") in new stack
    -- Executing [dstring@macro-dial-one:13] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(DIALSTATUS=CHANUNAVAIL)") in new stack
    -- Executing [dstring@macro-dial-one:14] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?skipset") in new stack
    -- Executing [dstring@macro-dial-one:15] Set("PJSIP/Provider_SIP_Trunk-00000191", "DSTRING=PJSIP/2440/sip:[email protected]:5953&") in new stack
    -- Executing [dstring@macro-dial-one:16] Set("PJSIP/Provider_SIP_Trunk-00000191", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:17] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:18] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:19] Set("PJSIP/Provider_SIP_Trunk-00000191", "DSTRING=PJSIP/2440/sip:[email protected]:5953") in new stack
    -- Executing [dstring@macro-dial-one:20] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [s@macro-dial-one:29] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:30] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:31] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("PJSIP/Provider_SIP_Trunk-00000191", "DB(CALLTRACE/2440)=+4822669955") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [s@macro-dial-one:32] Set("PJSIP/Provider_SIP_Trunk-00000191", "D_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-dial-one:33] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [s@macro-dial-one:34] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Blind Transfer: , Attended Transfer: , User: , Alert Info: ") in new stack
    -- Executing [s@macro-dial-one:35] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:36] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:37] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:38] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
    -- Executing [s@macro-dial-one:39] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
    -- Executing [s@macro-dial-one:40] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [s@macro-dial-one:41] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "1?Set(CHANNEL(musicclass)=none)") in new stack
    -- Executing [s@macro-dial-one:42] GosubIf("PJSIP/Provider_SIP_Trunk-00000191", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:43] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:44] Set("PJSIP/Provider_SIP_Trunk-00000191", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:45] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:46] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?godial") in new stack
    -- Goto (macro-dial-one,s,51)
    -- Executing [s@macro-dial-one:51] Macro("PJSIP/Provider_SIP_Trunk-00000191", "dialout-one-predial-hook,") in new stack
    -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [s@macro-dial-one:52] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "1?Set(D_OPTIONS=HhtrI)") in new stack
    -- Executing [s@macro-dial-one:53] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(CWRING=r(callwaiting)):Set(CWRING=)") in new stack
    -- Executing [s@macro-dial-one:54] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
    -- Executing [s@macro-dial-one:55] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(D_OPTIONS=HhtrIg)") in new stack
    -- Executing [s@macro-dial-one:56] Dial("PJSIP/Provider_SIP_Trunk-00000191", "PJSIP/2440/sip:[email protected]:5953,90,HhtrIb(func-apply-sipheaders^s^1)") in new stack
    -- PJSIP/2440-00000192 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/2440-00000192", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
    -- Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/2440-00000192", "Applying SIP Headers to channel PJSIP/2440-00000192") in new stack
    -- Executing [s@func-apply-sipheaders:3] Set("PJSIP/2440-00000192", "TECH=PJSIP") in new stack
    -- Executing [s@func-apply-sipheaders:4] Set("PJSIP/2440-00000192", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:5] While("PJSIP/2440-00000192", "0") in new stack
    -- Jumping to priority 13
    -- Executing [s@func-apply-sipheaders:14] Return("PJSIP/2440-00000192", "") in new stack
  == Spawn extension (from-internal, 2440, 1) exited non-zero on 'PJSIP/2440-00000192'
    -- PJSIP/2440-00000192 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called PJSIP/2440/sip:[email protected]:5953
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio CoS mark 5
    -- Connected line update to PJSIP/Provider_SIP_Trunk-00000191 prevented.
       > 0x7fa1a8141170 -- Strict RTP switching to RTP target address 10.9.10.10:25600 as source
    -- PJSIP/2440-00000192 is ringing
       > 0x7fa1a8141170 -- Strict RTP learning complete - Locking on source address 10.9.10.10:25600
    -- PJSIP/2440-00000192 Internal Gosub(crm-hangup,s,1) start
    -- Executing [s@crm-hangup:1] NoOp("PJSIP/2440-00000192", "Sending Hangup to CRM") in new stack
    -- Executing [s@crm-hangup:2] NoOp("PJSIP/2440-00000192", "HANGUP CAUSE: 0") in new stack
    -- Executing [s@crm-hangup:3] ExecIf("PJSIP/2440-00000192", "0?Set(__CRM_VOICEMAIL=)") in new stack
    -- Executing [s@crm-hangup:4] NoOp("PJSIP/2440-00000192", "MASTER CHANNEL: 1725876673.806 = 1725876672.805") in new stack
    -- Executing [s@crm-hangup:5] GotoIf("PJSIP/2440-00000192", "1?return") in new stack
    -- Goto (crm-hangup,s,8)
    -- Executing [s@crm-hangup:8] Return("PJSIP/2440-00000192", "") in new stack
  == Spawn extension (from-internal, 2440, 1) exited non-zero on 'PJSIP/2440-00000192'
    -- PJSIP/2440-00000192 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
  == Spawn extension (macro-dial-one, s, 56) exited non-zero on 'PJSIP/Provider_SIP_Trunk-00000191' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 46) exited non-zero on 'PJSIP/Provider_SIP_Trunk-00000191' in macro 'exten-vm'
  == Spawn extension (ext-local, 2440, 3) exited non-zero on 'PJSIP/Provider_SIP_Trunk-00000191'
    -- Executing [h@ext-local:1] Macro("PJSIP/Provider_SIP_Trunk-00000191", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/Provider_SIP_Trunk-00000191' in macro 'hangupcall'
  == Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/Provider_SIP_Trunk-00000191'
    -- PJSIP/Provider_SIP_Trunk-00000191 Internal Gosub(crm-hangup,s,1) start
    -- Executing [s@crm-hangup:1] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "Sending Hangup to CRM") in new stack
    -- Executing [s@crm-hangup:2] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "HANGUP CAUSE: 16") in new stack
    -- Executing [s@crm-hangup:3] ExecIf("PJSIP/Provider_SIP_Trunk-00000191", "0?Set(__CRM_VOICEMAIL=)") in new stack
    -- Executing [s@crm-hangup:4] NoOp("PJSIP/Provider_SIP_Trunk-00000191", "MASTER CHANNEL: 1725876672.805 = 1725876672.805") in new stack
    -- Executing [s@crm-hangup:5] GotoIf("PJSIP/Provider_SIP_Trunk-00000191", "0?return") in new stack
    -- Executing [s@crm-hangup:6] Set("PJSIP/Provider_SIP_Trunk-00000191", "__CRM_HANGUP=1") in new stack
    -- Executing [s@crm-hangup:7] AGI("PJSIP/Provider_SIP_Trunk-00000191", "agi://127.0.0.1/sangomacrm.agi") in new stack
    -- <PJSIP/Provider_SIP_Trunk-00000191>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
    -- Executing [s@crm-hangup:8] Return("PJSIP/Provider_SIP_Trunk-00000191", "") in new stack
  == Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/Provider_SIP_Trunk-00000191'
    -- PJSIP/Provider_SIP_Trunk-00000191 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

system · October 10, 2024, 10:32am

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.